#include <audio_device_defines.h>
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| | AudioParameters () |
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| | AudioParameters (int sample_rate, size_t channels, size_t frames_per_buffer) |
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| void | reset (int sample_rate, size_t channels, size_t frames_per_buffer) |
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| size_t | bits_per_sample () const |
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| void | reset (int sample_rate, size_t channels, double ms_per_buffer) |
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| void | reset (int sample_rate, size_t channels) |
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| int | sample_rate () const |
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| size_t | channels () const |
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| size_t | frames_per_buffer () const |
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| size_t | frames_per_10ms_buffer () const |
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| size_t | GetBytesPerFrame () const |
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| size_t | GetBytesPerBuffer () const |
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| bool | is_valid () const |
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| bool | is_complete () const |
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| size_t | GetBytesPer10msBuffer () const |
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| double | GetBufferSizeInMilliseconds () const |
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| double | GetBufferSizeInSeconds () const |
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| | AudioParameters () |
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| | AudioParameters (int sample_rate, size_t channels, size_t frames_per_buffer) |
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| void | reset (int sample_rate, size_t channels, size_t frames_per_buffer) |
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| size_t | bits_per_sample () const |
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| void | reset (int sample_rate, size_t channels, double ms_per_buffer) |
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| void | reset (int sample_rate, size_t channels) |
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| int | sample_rate () const |
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| size_t | channels () const |
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| size_t | frames_per_buffer () const |
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| size_t | frames_per_10ms_buffer () const |
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| size_t | GetBytesPerFrame () const |
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| size_t | GetBytesPerBuffer () const |
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| bool | is_valid () const |
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| bool | is_complete () const |
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| size_t | GetBytesPer10msBuffer () const |
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| double | GetBufferSizeInMilliseconds () const |
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| double | GetBufferSizeInSeconds () const |
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◆ AudioParameters() [1/4]
| webrtc::AudioParameters::AudioParameters |
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inline |
◆ AudioParameters() [2/4]
| webrtc::AudioParameters::AudioParameters |
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int |
sample_rate, |
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size_t |
channels, |
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size_t |
frames_per_buffer |
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) |
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inline |
◆ AudioParameters() [3/4]
| webrtc::AudioParameters::AudioParameters |
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inline |
◆ AudioParameters() [4/4]
| webrtc::AudioParameters::AudioParameters |
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int |
sample_rate, |
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size_t |
channels, |
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size_t |
frames_per_buffer |
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) |
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inline |
◆ bits_per_sample() [1/2]
| size_t webrtc::AudioParameters::bits_per_sample |
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const |
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inline |
◆ bits_per_sample() [2/2]
| size_t webrtc::AudioParameters::bits_per_sample |
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const |
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inline |
◆ channels() [1/2]
| size_t webrtc::AudioParameters::channels |
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const |
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inline |
◆ channels() [2/2]
| size_t webrtc::AudioParameters::channels |
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const |
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inline |
◆ frames_per_10ms_buffer() [1/2]
| size_t webrtc::AudioParameters::frames_per_10ms_buffer |
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const |
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inline |
◆ frames_per_10ms_buffer() [2/2]
| size_t webrtc::AudioParameters::frames_per_10ms_buffer |
( |
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const |
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inline |
◆ frames_per_buffer() [1/2]
| size_t webrtc::AudioParameters::frames_per_buffer |
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const |
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inline |
◆ frames_per_buffer() [2/2]
| size_t webrtc::AudioParameters::frames_per_buffer |
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const |
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inline |
◆ GetBufferSizeInMilliseconds() [1/2]
| double webrtc::AudioParameters::GetBufferSizeInMilliseconds |
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const |
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inline |
◆ GetBufferSizeInMilliseconds() [2/2]
| double webrtc::AudioParameters::GetBufferSizeInMilliseconds |
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const |
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inline |
◆ GetBufferSizeInSeconds() [1/2]
| double webrtc::AudioParameters::GetBufferSizeInSeconds |
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const |
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inline |
◆ GetBufferSizeInSeconds() [2/2]
| double webrtc::AudioParameters::GetBufferSizeInSeconds |
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const |
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inline |
◆ GetBytesPer10msBuffer() [1/2]
| size_t webrtc::AudioParameters::GetBytesPer10msBuffer |
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const |
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inline |
◆ GetBytesPer10msBuffer() [2/2]
| size_t webrtc::AudioParameters::GetBytesPer10msBuffer |
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const |
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inline |
◆ GetBytesPerBuffer() [1/2]
| size_t webrtc::AudioParameters::GetBytesPerBuffer |
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const |
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inline |
◆ GetBytesPerBuffer() [2/2]
| size_t webrtc::AudioParameters::GetBytesPerBuffer |
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const |
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◆ GetBytesPerFrame() [1/2]
| size_t webrtc::AudioParameters::GetBytesPerFrame |
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const |
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inline |
◆ GetBytesPerFrame() [2/2]
| size_t webrtc::AudioParameters::GetBytesPerFrame |
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const |
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inline |
◆ is_complete() [1/2]
| bool webrtc::AudioParameters::is_complete |
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const |
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◆ is_complete() [2/2]
| bool webrtc::AudioParameters::is_complete |
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const |
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◆ is_valid() [1/2]
| bool webrtc::AudioParameters::is_valid |
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const |
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inline |
◆ is_valid() [2/2]
| bool webrtc::AudioParameters::is_valid |
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const |
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inline |
◆ reset() [1/6]
| void webrtc::AudioParameters::reset |
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int |
sample_rate, |
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size_t |
channels, |
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size_t |
frames_per_buffer |
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) |
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inline |
◆ reset() [2/6]
| void webrtc::AudioParameters::reset |
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int |
sample_rate, |
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size_t |
channels, |
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size_t |
frames_per_buffer |
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) |
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inline |
◆ reset() [3/6]
| void webrtc::AudioParameters::reset |
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int |
sample_rate, |
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size_t |
channels, |
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double |
ms_per_buffer |
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) |
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inline |
◆ reset() [4/6]
| void webrtc::AudioParameters::reset |
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int |
sample_rate, |
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size_t |
channels, |
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double |
ms_per_buffer |
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) |
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inline |
◆ reset() [5/6]
| void webrtc::AudioParameters::reset |
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int |
sample_rate, |
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size_t |
channels |
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) |
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inline |
◆ reset() [6/6]
| void webrtc::AudioParameters::reset |
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int |
sample_rate, |
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size_t |
channels |
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) |
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inline |
◆ sample_rate() [1/2]
| int webrtc::AudioParameters::sample_rate |
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const |
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inline |
◆ sample_rate() [2/2]
| int webrtc::AudioParameters::sample_rate |
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const |
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inline |
◆ kBitsPerSample
| static const size_t webrtc::AudioParameters::kBitsPerSample = 16 |
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static |
The documentation for this class was generated from the following file:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_device/include/audio_device_defines.h
